Real-time communication (RTC) is an integrated communication medium based on an upcoming W3C standard that makes use of other components like HTML5/JavaScript, the iLBC audio codec, and the VP8 video codec. It is has been adopted by many major browsers.
Despite the many advances in RTC and VoIP (Voice over IP) technology, there remains a technical problem for an objective way to directly assess a RTC and VOIP applications in terms of its quality for conversational uses. Conventional evaluation techniques for such applications rely upon assessments that measure properties such as latency and jitter. Such conventional assessment techniques may not reliably indicate call quality as perceived by a human listener. A more reliable assessment tool would be helpful for ensuring quality of service (QoS) and may also prove useful for the further development of RTC and VOIP applications.
Existing techniques for assessing call quality are technically deficient for lacking certain important features. For example, some existing approaches are subjective rather than objective, while others are designed for indirect evaluation, relying on features of the application and network under test, rather than real signals, leading to less accurate results. In addition, other existing techniques are not well-suited for conversational uses. While such approaches may be adequate for one-way communication, the measures they produce do not adequately reflect factors that affect human perception of call quality during conversations conducted via RTC or VoIP.